Multi-microphone robust noise suppression

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. application Ser. No. 12/832,920 (now U.S. Pat. No. 8,538,035, issued Sep. 17, 2013), filed Jul. 8, 2010, which claims the benefit of U.S. Provisional Application Ser. No. 61/329,322, filed Apr. 29, 2010. This application is related to U.S. patent application Ser. No. 12/832,901, filed Jul. 8, 2010. The disclosures of the aforementioned applications are incorporated herein by reference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates generally to audio processing, and more particularly to a noise suppression processing of an audio signal.

2. Description of Related Art

Currently, there are many methods for reducing background noise in an adverse audio environment. A stationary noise suppression system suppresses stationary noise, by either a fixed or varying number of dB. A fixed suppression system suppresses stationary or non-stationary noise by a fixed number of dB. The shortcoming of the stationary noise suppressor is that non-stationary noise will not be suppressed, whereas the shortcoming of the fixed suppression system is that it must suppress noise by a conservative level in order to avoid speech distortion at low signal-to-noise ratios (SNR).

Another form of noise suppression is dynamic noise suppression. A common type of dynamic noise suppression systems is based on SNR. The SNR may be used to determine a suppression value. Unfortunately, SNR by itself is not a very good predictor of speech distortion due to the presence of different noise types in the audio environment. Typically, speech energy, over a given period of time, will include a word, a pause, a word, a pause, and so forth. Additionally, stationary and dynamic noises may be present in the audio environment. The SNR averages all of these stationary and non-stationary speech and noise components. There is no consideration in the determination of the SNR of the characteristics of the noise signal—only the overall level of noise.

To overcome the shortcomings of the prior art, there is a need for an improved noise suppression system for processing audio signals.

SUMMARY OF THE INVENTION

The present technology provides a robust noise suppression system which may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

An embodiment includes a system for performing noise reduction in an audio signal may include a memory. A frequency analysis module stored in the memory and executed by a processor may generate sub-band signals in a cochlea domain from time domain acoustic signals. A noise cancellation module stored in the memory and executed by a processor may cancel at least a portion of the sub-band signals. A modifier module stored in the memory and executed by a processor may suppress a noise component or an echo component in the modified sub-band signals. A reconstructor module stored in the memory and executed by a processor may reconstruct a modified time domain signal from the component suppressed sub-band signals provided by the modifier module.

Noise reduction may also be performed as a process performed by a machine with a processor and memory. Additionally, a computer readable storage medium may be implemented in which a program is embodied, the program being executable by a processor to perform a method for reducing noise in an audio signal.

BRIEF DESCRIPTION OF THE DRAWINGS

<figref idref="DRAWINGS">FIG. 1</figref> is an illustration of an environment in which embodiments of the present technology may be used.

<figref idref="DRAWINGS">FIG. 2</figref> is a block diagram of an exemplary audio device.

<figref idref="DRAWINGS">FIG. 3</figref> is a block diagram of an exemplary audio processing system.

<figref idref="DRAWINGS">FIG. 4</figref> is a flowchart of an exemplary method for performing noise reduction for an acoustic signal.

<figref idref="DRAWINGS">FIG. 5</figref> is a flowchart of an exemplary method for extracting features from audio signals.

DETAILED DESCRIPTION OF THE INVENTION

The present technology provides a robust noise suppression system which may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain. The present technology is both a dynamic and non-stationary noise suppression system, and provides a “perceptually optimal” amount of noise suppression based upon the characteristics of the noise and use case.

Performing noise (and echo) reduction via a combination of noise cancellation and noise suppression allows for flexibility in audio device design. In particular, a combination of subtractive and multiplicative stages is advantageous because it allows for both flexibility of microphone placement on an audio device and use case (e.g. close-talk/far-talk) whilst optimizing the overall tradeoff of voice quality vs. noise suppression. The microphones may be positioned within four centimeters of each other for a “close microphone” configuration” or greater than four centimeters apart for a “spread microphone” configuration, or a combination of configurations with greater than two microphones.

<figref idref="DRAWINGS">FIG. 1</figref> is an illustration of an environment in which embodiments of the present technology may be used. A user may act as an audio (speech) source <b>102</b> to an audio device <b>104</b>. The exemplary audio device <b>104</b> includes two microphones: a primary microphone <b>106</b> relative to the audio source <b>102</b> and a secondary microphone <b>108</b> located a distance away from the primary microphone <b>106</b>. Alternatively, the audio device <b>104</b> may include a single microphone. In yet other embodiments, the audio device <b>104</b> may include more than two microphones, such as for example three, four, five, six, seven, eight, nine, ten or even more microphones.

The primary microphone <b>106</b> and secondary microphone <b>108</b> may be omni-directional microphones. Alternatively embodiments may utilize other forms of microphones or acoustic sensors, such as directional microphones.

While the microphones <b>106</b> and <b>108</b> receive sound (i.e. acoustic signals) from the audio source <b>102</b>, the microphones <b>106</b> and <b>108</b> also pick up noise <b>112</b>. Although the noise <b>112</b> is shown coming from a single location in <figref idref="DRAWINGS">FIG. 1</figref>, the noise <b>112</b> may include any sounds from one or more locations that differ from the location of audio source <b>102</b>, and may include reverberations and echoes. The noise <b>112</b> may be stationary, non-stationary, and/or a combination of both stationary and non-stationary noise.

Some embodiments may utilize level differences (e.g. energy differences) between the acoustic signals received by the two microphones <b>106</b> and <b>108</b>. Because the primary microphone <b>106</b> is much closer to the audio source <b>102</b> than the secondary microphone <b>108</b> in a close-talk use case, the intensity level is higher for the primary microphone <b>106</b>, resulting in a larger energy level received by the primary microphone <b>106</b> during a speech/voice segment, for example.

The level difference may then be used to discriminate speech and noise in the time-frequency domain. Further embodiments may use a combination of energy level differences and time delays to discriminate speech. Based on binaural cue encoding, speech signal extraction or speech enhancement may be performed.

<figref idref="DRAWINGS">FIG. 2</figref> is a block diagram of an exemplary audio device <b>104</b>. In the illustrated embodiment, the audio device <b>104</b> includes a receiver <b>200</b>, a processor <b>202</b>, the primary microphone <b>106</b>, an optional secondary microphone <b>108</b>, an audio processing system <b>210</b>, and an output device <b>206</b>. The audio device <b>104</b> may include further or other components necessary for audio device <b>104</b> operations. Similarly, the audio device <b>104</b> may include fewer components that perform similar or equivalent functions to those depicted in <figref idref="DRAWINGS">FIG. 2</figref>.

Processor <b>202</b> may execute instructions and modules stored in a memory (not illustrated in <figref idref="DRAWINGS">FIG. 2</figref>) in the audio device <b>104</b> to perform functionality described herein, including noise reduction for an acoustic signal. Processor <b>202</b> may include hardware and software implemented as a processing unit, which may process floating point operations and other operations for the processor <b>202</b>.

The exemplary receiver <b>200</b> is an acoustic sensor configured to receive a signal from a communications network. In some embodiments, the receiver <b>200</b> may include an antenna device. The signal may then be forwarded to the audio processing system <b>210</b> to reduce noise using the techniques described herein, and provide an audio signal to the output device <b>206</b>. The present technology may be used in one or both of the transmit and receive paths of the audio device <b>104</b>.

The audio processing system <b>210</b> is configured to receive the acoustic signals from an acoustic source via the primary microphone <b>106</b> and secondary microphone <b>108</b> and process the acoustic signals. Processing may include performing noise reduction within an acoustic signal. The audio processing system <b>210</b> is discussed in more detail below. The primary and secondary microphones <b>106</b>, <b>108</b> may be spaced a distance apart in order to allow for detecting an energy level difference, time difference or phase difference between them. The acoustic signals received by primary microphone <b>106</b> and secondary microphone <b>108</b> may be converted into electrical signals (i.e. a primary electrical signal and a secondary electrical signal). The electrical signals may themselves be converted by an analog-to-digital converter (not shown) into digital signals for processing in accordance with some embodiments. In order to differentiate the acoustic signals for clarity purposes, the acoustic signal received by the primary microphone <b>106</b> is herein referred to as the primary acoustic signal, while the acoustic signal received from by the secondary microphone <b>108</b> is herein referred to as the secondary acoustic signal. The primary acoustic signal and the secondary acoustic signal may be processed by the audio processing system <b>210</b> to produce a signal with an improved signal-to-noise ratio. It should be noted that embodiments of the technology described herein may be practiced utilizing only the primary microphone <b>106</b>.

The output device <b>206</b> is any device which provides an audio output to the user. For example, the output device <b>206</b> may include a speaker, an earpiece of a headset or handset, or a speaker on a conference device.

In various embodiments, where the primary and secondary microphones are omni-directional microphones that are closely-spaced (e.g., 1-2 cm apart), a beamforming technique may be used to simulate forwards-facing and backwards-facing directional microphones. The level difference may be used to discriminate speech and noise in the time-frequency domain which can be used in noise reduction.

<figref idref="DRAWINGS">FIG. 3</figref> is a block diagram of an exemplary audio processing system <b>210</b> for performing noise reduction as described herein. In exemplary embodiments, the audio processing system <b>210</b> is embodied within a memory device within audio device <b>104</b>. The audio processing system <b>210</b> may include a frequency analysis module <b>302</b>, a feature extraction module <b>304</b>, a source inference engine module <b>306</b>, mask generator module <b>308</b>, noise canceller module <b>310</b>, modifier module <b>312</b>, and reconstructor module <b>314</b>. Audio processing system <b>210</b> may include more or fewer components than illustrated in <figref idref="DRAWINGS">FIG. 3</figref>, and the functionality of modules may be combined or expanded into fewer or additional modules. Exemplary lines of communication are illustrated between various modules of <figref idref="DRAWINGS">FIG. 3</figref>, and in other figures herein. The lines of communication are not intended to limit which modules are communicatively coupled with others, nor are they intended to limit the number of and type of signals communicated between modules.

In operation, acoustic signals received from the primary microphone <b>106</b> and second microphone <b>108</b> are converted to electrical signals, and the electrical signals are processed through frequency analysis module <b>302</b>. The acoustic signals may be pre-processed in the time domain before being processed by frequency analysis module <b>302</b>. Time domain pre-processing may include applying input limiter gains, speech time stretching, and filtering using an FIR or IIR filter.

The frequency analysis module <b>302</b> takes the acoustic signals and mimics the frequency analysis of the cochlea (e.g., cochlear domain), simulated by a filter bank. The frequency analysis module <b>302</b> separates each of the primary and secondary acoustic signals into two or more frequency sub-band signals. A sub-band signal is the result of a filtering operation on an input signal, where the bandwidth of the filter is narrower than the bandwidth of the signal received by the frequency analysis module <b>302</b>. The filter bank may be implemented by a series of cascaded, complex-valued, first-order IIR filters. Alternatively, other filters such as short-time Fourier transform (STFT), sub-band filter banks, modulated complex lapped transforms, cochlear models, wavelets, etc., can be used for the frequency analysis and synthesis. The samples of the frequency sub-band signals may be grouped sequentially into time frames (e.g. over a predetermined period of time). For example, the length of a frame may be 4 ms, 8 ms, or some other length of time. In some embodiments there may be no frame at all. The results may include sub-band signals in a fast cochlea transform (FCT) domain.

The sub-band frame signals are provided from frequency analysis module <b>302</b> to an analysis path sub-system <b>320</b> and a signal path sub-system <b>330</b>. The analysis path sub-system <b>320</b> may process the signal to identify signal features, distinguish between speech components and noise components of the sub-band signals, and generate a signal modifier. The signal path sub-system <b>330</b> is responsible for modifying sub-band signals of the primary acoustic signal by reducing noise in the sub-band signals. Noise reduction can include applying a modifier, such as a multiplicative gain mask generated in the analysis path sub-system <b>320</b>, or by subtracting components from the sub-band signals. The noise reduction may reduce noise and preserve the desired speech components in the sub-band signals.

Signal path sub-system <b>330</b> includes noise canceller module <b>310</b> and modifier module <b>312</b>. Noise canceller module <b>310</b> receives sub-band frame signals from frequency analysis module <b>302</b>. Noise canceller module <b>310</b> may subtract (e.g., cancel) a noise component from one or more sub-band signals of the primary acoustic signal. As such, noise canceller module <b>310</b> may output sub-band estimates of noise components in the primary signal and sub-band estimates of speech components in the form of noise-subtracted sub-band signals.

Noise canceller module <b>310</b> may provide noise cancellation, for example in systems with two-microphone configurations, based on source location by means of a subtractive algorithm. Noise canceller module <b>310</b> may also provide echo cancellation and is intrinsically robust to loudspeaker and Rx path non-linearity. By performing noise and echo cancellation (e.g., subtracting components from a primary signal sub-band) with little or no voice quality degradation, noise canceller module <b>310</b> may increase the speech-to-noise ratio (SNR) in sub-band signals received from frequency analysis module <b>302</b> and provided to modifier module <b>312</b> and post filtering modules. The amount of noise cancellation performed may depend on the diffuseness of the noise source and the distance between microphones, both of which contribute towards the coherence of the noise between the microphones, with greater coherence resulting in better cancellation.

Noise canceller module <b>310</b> may be implemented in a variety of ways. In some embodiments, noise canceller module <b>310</b> may be implemented with a single null processing noise subtraction (NPNS) module. Alternatively, noise canceller module <b>310</b> may include two or more NPNS modules, which may be arranged for example in a cascaded fashion.

An example of noise cancellation performed in some embodiments by the noise canceller module <b>310</b> is disclosed in U.S. patent application Ser. No. 12/215,980, entitled “System and Method for Providing Noise Suppression Utilizing Null Processing Noise Subtraction,” filed Jun. 30, 2008, U.S. application Ser. No. 12/422,917, entitled “Adaptive Noise Cancellation,” filed Apr. 13, 2009, and U.S. application Ser. No. 12/693,998, entitled “Adaptive Noise Reduction Using Level Cues,” filed Jan. 26, 2010, the disclosures of which are each incorporated herein by reference.

The feature extraction module <b>304</b> of the analysis path sub-system <b>320</b> receives the sub-band frame signals derived from the primary and secondary acoustic signals provided by frequency analysis module <b>302</b> as well as the output of NPNS module <b>310</b>. Feature extraction module <b>304</b> computes frame energy estimations of the sub-band signals, inter-microphone level differences (ILD), inter-microphone time differences (ITD) and inter-microphones phase differences (IPD) between the primary acoustic signal and the secondary acoustic signal, self-noise estimates for the primary and second microphones, as well as other monaural or binaural features which may be utilized by other modules, such as pitch estimates and cross-correlations between microphone signals. The feature extraction module <b>304</b> may both provide inputs to and process outputs from NPNS module <b>310</b>.

Feature extraction module <b>304</b> may generate a null-processing inter-microphone level difference (NP-ILD). The NP-ILD may be used interchangeably in the present system with a raw ILD. A raw ILD between a primary and secondary microphone may be determined by an ILD module within feature extraction module <b>304</b>. The ILD computed by the ILD module in one embodiment may be represented mathematically by

<maths id="MATH-US-00001" num="00001"><math overflow="scroll"><mrow><mrow><mi>I</mi><mo>⁢</mo><mstyle><mspace width="0.3em" height="0.3ex" /></mstyle><mo>⁢</mo><mi>L</mi><mo>⁢</mo><mstyle><mspace width="0.3em" height="0.3ex" /></mstyle><mo>⁢</mo><mi>D</mi></mrow><mo>=</mo><msub><mrow><mo>⌈</mo><msub><mrow><mo>⌊</mo><mrow><mi>c</mi><mo>·</mo><mrow><msub><mi>log</mi><mn>2</mn></msub><mo>⁡</mo><mrow><mo>(</mo><mfrac><msub><mi>E</mi><mn>1</mn></msub><msub><mi>E</mi><mn>2</mn></msub></mfrac><mo>)</mo></mrow></mrow></mrow><mo>⌋</mo></mrow><mrow><mo>-</mo><mn>1</mn></mrow></msub><mo>⌉</mo></mrow><mrow><mo>+</mo><mn>1</mn></mrow></msub></mrow></math></maths><img file="US9438992B2_D0001.tif" />

where E1 and E2 are the energy outputs of the primary and secondary microphones <b>106</b>, <b>108</b>, respectively, computed in each sub-band signal over non-overlapping time intervals (“frames”). This equation describes the dB ILD normalized by a factor of c and limited to the range [−1, +1]. Thus, when the audio source <b>102</b> is close to the primary microphone <b>106</b> for E1 and there is no noise, ILD=1, but as more noise is added, the ILD will be reduced.

In some cases, where the distance between microphones is small with respect to the distance between the primary microphone and the mouth, raw ILD may not be useful to discriminate a source from a distracter, since both source and distracter may have roughly equal raw ILD. In order to avoid limitations regarding raw ILD used to discriminate a source from a distracter, outputs of noise canceller module <b>310</b> may be used to derive an ILD having a positive value for the speech signal and small or negative value for the noise components since these will be significantly attenuated at the output of the noise canceller module <b>310</b>. The ILD derived from the noise canceller module <b>310</b> outputs may be a Null Processing Inter-microphone Level Difference (NP-ILD), and represented mathematically by:

<maths id="MATH-US-00002" num="00002"><math overflow="scroll"><mrow><mrow><mrow><mi>N</mi><mo>⁢</mo><mstyle><mspace width="0.3em" height="0.3ex" /></mstyle><mo>⁢</mo><mi>P</mi></mrow><mo>-</mo><mrow><mi>I</mi><mo>⁢</mo><mstyle><mspace width="0.3em" height="0.3ex" /></mstyle><mo>⁢</mo><mi>L</mi><mo>⁢</mo><mstyle><mspace width="0.3em" height="0.3ex" /></mstyle><mo>⁢</mo><mi>D</mi></mrow></mrow><mo>=</mo><msub><mrow><mo>⌈</mo><msub><mrow><mo>⌊</mo><mrow><mi>c</mi><mo>·</mo><mrow><msub><mi>log</mi><mn>2</mn></msub><mo>⁡</mo><mrow><mo>(</mo><mfrac><msub><mi>E</mi><mi>NP</mi></msub><msub><mi>E</mi><mn>2</mn></msub></mfrac><mo>)</mo></mrow></mrow></mrow><mo>⌋</mo></mrow><mrow><mo>-</mo><mn>1</mn></mrow></msub><mo>⌉</mo></mrow><mrow><mo>+</mo><mn>1</mn></mrow></msub></mrow></math></maths><img file="US9438992B2_D0002.tif" />

NPNS module may provide noise cancelled sub-band signals to the ILD block in the feature extraction module <b>304</b>. Since the ILD may be determined as the ratio of the NPNS output signal energy to the secondary microphone energy, ILD is often interchangeable with Null Processing Inter-microphone Level Difference (NP-ILD). “Raw-ILD” may be used to disambiguate a case where the ILD is computed from the “raw” primary and secondary microphone signals.

Determining energy level estimates and inter-microphone level differences is discussed in more detail in U.S. patent application Ser. No. 11/343,524, entitled “System and Method for Utilizing Inter-Microphone Level Differences for Speech Enhancement”, which is incorporated by reference herein.

Source inference engine module <b>306</b> may process the frame energy estimations provided by feature extraction module <b>304</b> to compute noise estimates and derive models of the noise and speech in the sub-band signals. Source inference engine module <b>306</b> adaptively estimates attributes of the acoustic sources, such as their energy spectra of the output signal of the NPNS module <b>310</b>. The energy spectra attribute may be utilized to generate a multiplicative mask in mask generator module <b>308</b>.

The source inference engine module <b>306</b> may receive the NP-ILD from feature extraction module <b>304</b> and track the NP-ILD probability distributions or “clusters” of the target audio source <b>102</b>, background noise and optionally echo.

This information is then used, along with other auditory cues, to define classification boundaries between source and noise classes. The NP-ILD distributions of speech, noise and echo may vary over time due to changing environmental conditions, movement of the audio device <b>104</b>, position of the hand and/or face of the user, other objects relative to the audio device <b>104</b>, and other factors. The cluster tracker adapts to the time-varying NP-ILDs of the speech or noise source(s).

When ignoring echo, without any loss of generality, when the source and noise ILD distributions are non-overlapping, it is possible to specify a classification boundary or dominance threshold between the two distributions, such that the signal is classified as speech if the SNR is sufficiently positive or as noise if the SNR is sufficiently negative. This classification may be determined per sub-band and time-frame as a dominance mask, and output by a cluster tracker module to a noise estimator module within the source inference engine module <b>306</b>.

The cluster tracker may determine a global summary of acoustic features based, at least in part, on acoustic features derived from an acoustic signal, as well as an instantaneous global classification based on a global running estimate and the global summary of acoustic features. The global running estimates may be updated and an instantaneous local classification is derived based on at least the one or more acoustic features. Spectral energy classifications may then be determined based, at least in part, on the instantaneous local classification and the one or more acoustic features.

In some embodiments, the cluster tracker module classifies points in the energy spectrum as being speech or noise based on these local clusters and observations. As such, a local binary mask for each point in the energy spectrum is identified as either speech or noise.

The cluster tracker module may generate a noise/speech classification signal per sub-band and provide the classification to NPNS module <b>310</b>. In some embodiments, the classification is a control signal indicating the differentiation between noise and speech. Noise canceller module <b>310</b> may utilize the classification signals to estimate noise in received microphone signals. In some embodiments, the results of cluster tracker module may be forwarded to the noise estimate module within the source inference engine module <b>306</b>. In other words, a current noise estimate along with locations in the energy spectrum where the noise may be located are provided for processing a noise signal within audio processing system <b>210</b>.

An example of tracking clusters by a cluster tracker module is disclosed in U.S. patent application Ser. No. 12/004,897, entitled “System and Method for Adaptive Classification of Audio Sources,” filed on Dec. 21, 2007, the disclosure of which is incorporated herein by reference.

Source inference engine module <b>306</b> may include a noise estimate module which may receive a noise/speech classification control signal from the cluster tracker module and the output of noise canceller module <b>310</b> to estimate the noise N(t,w), wherein t is a point in time and W represents a frequency or sub-band. The noise estimate determined by noise estimate module is provided to mask generator module <b>308</b>. In some embodiments, mask generator module <b>308</b> receives the noise estimate output of noise canceller module <b>310</b> and an output of the cluster tracker module.

The noise estimate module in the source inference engine module <b>306</b> may include an NP-ILD noise estimator and a stationary noise estimator. The noise estimates can be combined, such as for example with a max( ) operation, so that the noise suppression performance resulting from the combined noise estimate is at least that of the individual noise estimates.

The NP-ILD noise estimate may be derived from the dominance mask and noise canceller module <b>310</b> output signal energy. When the dominance mask is 1 (indicating speech) in a particular sub-band, the noise estimate is frozen, and when the dominance mask is 0 (indicating noise) in a particular sub-band, the noise estimate is set equal to the NPNS output signal energy. The stationary noise estimate tracks components of the NPNS output signal that vary more slowly than speech typically does, and the main input to this module is the NPNS output energy.

The mask generator module <b>308</b> receives models of the sub-band speech components and noise components as estimated by the source inference engine module <b>306</b> and generates a multiplicative mask. The multiplicative mask is applied to the estimated noise subtracted sub-band signals provided by NPNS <b>310</b> to modifier <b>312</b>. The modifier module <b>312</b> multiplies the gain masks to the noise-subtracted sub-band signals of the primary acoustic signal output by the NPNS module <b>310</b>. Applying the mask reduces energy levels of noise components in the sub-band signals of the primary acoustic signal and results in noise reduction.

The multiplicative mask is defined by a Wiener filter and a voice quality optimized suppression system. The Wiener filter estimate may be based on the power spectral density of noise and a power spectral density of the primary acoustic signal. The Wiener filter derives a gain based on the noise estimate. The derived gain is used to generate an estimate of the theoretical MMSE of the clean speech signal given the noisy signal. To limit the amount of speech distortion as a result of the mask application, the Wiener gain may be limited at a lower end using a perceptually-derived gain lower bound

The values of the gain mask output from mask generator module <b>308</b> are time and sub-band signal dependent and optimize noise reduction on a per sub-band basis. The noise reduction may be subject to the constraint that the speech loss distortion complies with a tolerable threshold limit. The threshold limit may be based on many factors, such as for example a voice quality optimized suppression (VQOS) level. The VQOS level is an estimated maximum threshold level of speech loss distortion in the sub-band signal introduced by the noise reduction. The VQOS is tunable and takes into account the properties of the sub-band signal, and provides full design flexibility for system and acoustic designers. A lower bound for the amount of noise reduction performed in a sub-band signal is determined subject to the VQOS threshold, thereby limiting the amount of speech loss distortion of the sub-band signal. As a result, a large amount of noise reduction may be performed in a sub-band signal when possible, and the noise reduction may be smaller when conditions such as unacceptably high speech loss distortion do not allow for the large amount of noise reduction.

In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level, which may be fixed or slowly time-varying. In some embodiments, the residual noise target level is the same for each sub-band signal, in other embodiments it may vary across sub-bands. Such a target level may be a level at which the noise component ceases to be audible or perceptible, below a self-noise level of a microphone used to capture the primary acoustic signal, or below a noise gate of a component on a baseband chip or of an internal noise gate within a system implementing the noise reduction techniques.

Modifier module <b>312</b> receives the signal path cochlear samples from noise canceller module <b>310</b> and applies a gain mask received from mask generator <b>308</b> to the received samples. The signal path cochlear samples may include the noise subtracted sub-band signals for the primary acoustic signal. The mask provided by the Weiner filter estimation may vary quickly, such as from frame to frame, and noise and speech estimates may vary between frames. To help address the variance, the upwards and downwards temporal slew rates of the mask may be constrained to within reasonable limits by modifier <b>312</b>. The mask may be interpolated from the frame rate to the sample rate using simple linear interpolation, and applied to the sub-band signals by multiplicative noise suppression. Modifier module <b>312</b> may output masked frequency sub-band signals.

Reconstructor module <b>314</b> may convert the masked frequency sub-band signals from the cochlea domain back into the time domain. The conversion may include adding the masked frequency sub-band signals and phase shifted signals. Alternatively, the conversion may include multiplying the masked frequency sub-band signals with an inverse frequency of the cochlea channels. Once conversion to the time domain is completed, the synthesized acoustic signal may be output to the user via output device <b>206</b> and/or provided to a codec for encoding.

In some embodiments, additional post-processing of the synthesized time domain acoustic signal may be performed. For example, comfort noise generated by a comfort noise generator may be added to the synthesized acoustic signal prior to providing the signal to the user. Comfort noise may be a uniform constant noise that is not usually discernible to a listener (e.g., pink noise). This comfort noise may be added to the synthesized acoustic signal to enforce a threshold of audibility and to mask low-level non-stationary output noise components. In some embodiments, the comfort noise level may be chosen to be just above a threshold of audibility and may be settable by a user. In some embodiments, the mask generator module <b>308</b> may have access to the level of comfort noise in order to generate gain masks that will suppress the noise to a level at or below the comfort noise.

The system of <figref idref="DRAWINGS">FIG. 3</figref> may process several types of signals received by an audio device. The system may be applied to acoustic signals received via one or more microphones. The system may also process signals, such as a digital Rx signal, received through an antenna or other connection.

<figref idref="DRAWINGS">FIGS. 4 and 5</figref> include flowcharts of exemplary methods for performing the present technology. Each step of <figref idref="DRAWINGS">FIGS. 4 and 5</figref> may be performed in any order, and the methods of <figref idref="DRAWINGS">FIGS. 4 and 5</figref> may each include additional or fewer steps than those illustrated.

<figref idref="DRAWINGS">FIG. 4</figref> is a flowchart of an exemplary method for performing noise reduction for an acoustic signal. Microphone acoustic signals may be received at step <b>405</b>. The acoustic signals received by microphones <b>106</b> and <b>108</b> may each include at least a portion of speech and noise. Pre-processing may be performed on the acoustic signals at step <b>410</b>. The pre-processing may include applying a gain, equalization and other signal processing to the acoustic signals.

Sub-band signals are generated in a cochlea domain at step <b>415</b>. The sub-band signals may be generated from time domain signals using a cascade of complex filters.

Feature extraction is performed at step <b>420</b>. The feature extraction may extract features from the sub-band signals that are used to cancel a noise component, infer whether a sub-band has noise or echo, and generate a mask. Performing feature extraction is discussed in more detail with respect to <figref idref="DRAWINGS">FIG. 5</figref>.

Noise cancellation is performed at step <b>425</b>. The noise cancellation can be performed by NPNS module <b>310</b> on one or more sub-band signals received from frequency analysis module <b>302</b>. Noise cancellation may include subtracting a noise component from a primary acoustic signal sub-band. In some embodiments, an echo component may be cancelled from a primary acoustic signal sub-band. The noise-cancelled (or echo-cancelled) signal may be provided to feature extraction module <b>304</b> to determine a noise component energy estimate and to source inference engine <b>306</b>.

A noise estimate, echo estimate, and speech estimate may be determined for sub-bands at step <b>430</b>. Each estimate may be determined for each sub-band in an acoustic signal and for each frame in the acoustic audio signal. The echo may be determined at least in part from an Rx signal received by source inference engine <b>306</b>. The inference as to whether a sub-band within a particular time frame is determined to be noise, speech or echo is provided to mask generator module <b>308</b>.

A mask is generated at step <b>435</b>. The mask may be generated by mask generator <b>308</b>. A mask may be generated and applied to each sub-band during each frame based on a determination as to whether the particular sub-band is determined to be noise, speech or echo. The mask may be generated based on voice quality optimized suppression—a level of suppression determined to be optimized for a particular level of voice distortion. The mask may then be applied to a sub-band at step <b>440</b>. The mask may be applied by modifier <b>312</b> to the sub-band signals output by NPNS <b>310</b>. The mask may be interpolated from frame rate to sample rate by modifier <b>312</b>.

A time domain signal is reconstructed from sub-band signals at step <b>445</b>. The time band signal may be reconstructed by applying a series of delays and complex multiply operations to the sub-band signals by reconstructor module <b>314</b>. Post processing may then be performed on the reconstructed time domain signal at step <b>450</b>. The post processing may be performed by a post processor and may include applying an output limiter to the reconstructed signal, applying an automatic gain control, and other post-processing. The reconstructed output signal may then be output at step <b>455</b>.

<figref idref="DRAWINGS">FIG. 5</figref> is a flowchart of an exemplary method for extracting features from audio signals. The method of <figref idref="DRAWINGS">FIG. 5</figref> may provide more detail for step <b>420</b> of the method of <figref idref="DRAWINGS">FIG. 4</figref>. Sub-band signals are received at step <b>505</b>. Feature extraction module <b>304</b> may receive sub-band signals from frequency analysis module <b>302</b> and output signals from noise canceller module <b>310</b>. Second order statistics, such as for example sub-band energy levels, are determined at step <b>510</b>. The energy sub-band levels may be determined for each sub-band for each frame. Cross correlations between microphones and autocorrelations of microphone signals may be calculated at step <b>515</b>. An inter-microphone level difference (ILD) is determined at step <b>520</b>. A null processing inter-microphone level difference (NP-ILD) is determined at step <b>525</b>. Both the ILD and the NP-ILD are determined at least in part from the sub-band signal energy and the noise estimate energy. The extracted features are then utilized by the audio processing system in reducing the noise in sub-band signals.

The above described modules, including those discussed with respect to <figref idref="DRAWINGS">FIG. 3</figref>, may include instructions stored in a storage media such as a machine readable medium (e.g., computer readable medium). These instructions may be retrieved and executed by the processor <b>202</b> to perform the functionality discussed herein. Some examples of instructions include software, program code, and firmware. Some examples of storage media include memory devices and integrated circuits.

While the present invention is disclosed by reference to the preferred embodiments and examples detailed above, it is to be understood that these examples are intended in an illustrative rather than a limiting sense. It is contemplated that modifications and combinations will readily occur to those skilled in the art, which modifications and combinations will be within the spirit of the invention and the scope of the following claims.